Voice connections in telecommunication networks have in the past normally been set up on a connection-oriented basis. To do this, one line is provided exclusively for signal transmission between two communication endpoints and is, so to speak, reserved for this voice connection. In this context, the literature frequently refers to line-switching or line-oriented telecommunication.
With the arrival of packet-oriented data networks, such as the Internet, telecommunication is possible more cost-effectively than by using line-oriented telecommunication. This is due, in particular, to the capability to use the available connection resources better, since the resources that exist in a telecommunication network, in particular transmission capacities, can be used far more efficiently via packet-oriented transmission than is possible in the case of line-oriented transmission with an assured line capacity.
VoF (short for “Voice over Frame Relay”) or VoIP (short for “Voice over IP”) are known, by way of example, as voice packet-oriented transmission methods. VoIP technology, in particular, is predicted to be of major importance for future voice communication.
However, the transmission of voice data via packet-oriented transmission methods is subject to the problem that the transmission bandwidth available for a voice connection fluctuates as a function of the load level in a data network. Normally, this leads to delays (frequently referred to as a delay or jitter in the literature) or even to gaps in the voice connection. In the worst case, the voice connection may even fail completely. The so-called QoS (short for “Quality of Service”) of such a voice connection is thus considerably worse than that of line-switching communication. In order to remedy these problems at least partially, voice compression methods, such as G.723.1, are used to reduce the bandwidth required for voice communication.
Since the Internet is frequently used as the data network for VoIP technology, despite the fact that it is not very suitable since the bandwidth available for voice communication fluctuates during most access procedures, it is particularly important to maintain a minimum bandwidth for a connection quality that is defined as the minimum. The routers which are used for setting up connections control the bandwidth on the basis of the current bandwidth demand for a voice connection. Specifically, this means that at least one new transmission channel is set up for a VoIP connection, depending on the currently required bandwidth.
However, interference can occur in the voice connection in this case since additional bandwidth is requested only when a demand occurs and, in consequence, the voice connection is subject to relatively major gaps and/or delays. The router makes its decision to request additional bandwidth on the basis of the routed data, that is to say only at a time at which additional bandwidth is already required. Thus, even with this method, a voice connection without any interference at all is impossible.
In order to explain this better, the following text refers to FIG. 4. Shown is an arrangement with a router for setting up connections between communication endpoints. Two transmission channels 52 and 54 are set up between a router 50, as a first communication end point, and a remote point 56, as the second communication end point. The remote point is a PPP interface (Point to Point Protocol), which allows the Internet protocol TCP/IP to be used via a telecommunication network. A control unit 58 includes a measurement unit 60 and a threshold value control unit 62. The measurement unit 60 measures the data throughput rate via the two transmission channels 52 and 54.
When a connection request occurs, the control unit 58 uses the measurement unit 60 to determine the data throughput rate and, if necessary, uses the threshold value control unit 62 to set up additional data channels for the requested connection. If an already existing connection requires additional bandwidth and requests this, then additional data channels are likewise set up although, in fact, the speech quality will be poor while the additional data channels are being set up. In some circumstances, the setting-up process may even occur at such a late state that the voice connection is interrupted for a certain period of time, and voice data is lost owing to the lack of bandwidth.
New methods have been proposed at the protocol level to solve these problems. One of these is an end-to-end Internet protocol from the IETF (Internet Engineering Task Force) and the company Cisco, which is referred to as RSVP (short for “Resource Reservation Setup Protocol”). In order to maintain a specific QoS for applications via the Internet, network resources, such as bandwidth, are reserved for a transmission. RSVP not only reserves resources before the transmission of data, but also adapts the transmission capacities dynamically. However, RSVP is a proprietary protocol which must be procured for all the components involved in a transmission. Furthermore, the RSVP protocol is highly complex, for which reason it is not yet widely used. Furthermore, the technical complexity for implementing the RSVP protocol is considerable.
The present invention is thus directed toward providing a method for adjusting the bandwidth of a connection between at least two connection end points in a data network, and an apparatus for carrying out the method, which ensure, even before transmission, that the bandwidth is sufficient for voice connections, and which can be used in conventional telecommunication networks without any additional protocol complexity.